[Asterisk-Users] Asterisk and Cisco 7960

[Andrew Joakimsen]

> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-
> admin at lists.digium.com] On Behalf Of Ben Wern
> Sent: Saturday, August 30, 2003 3:02 AM
> To: asterisk-users at lists.digium.com; asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] Asterisk and Cisco 7960
> 
> Andrew,
> 
> Thanks for your help!
No problem, that's what the list is for :)


> I did have the outgoing proxy set -- since I had FWD set up on line 1.
I
> removed all the FWD stuff, and the outgoing proxy. I altered the entry
to
> have the qualify, canreinvite, and nat lines and also altered the user
id
> to be a number. Now I'm able to call other local extensions, but I
can't
> call into the Cisco. But it's progress!
> 
> I can also call out to FWD, but audio drops after a few seconds. Don't
> even
> want to think about getting FWD calls back into the network.
Change your dial strings end in ,Tt) or ,Ttr)


> >exten => 1000,1,Dial(SIP/1000 at 1000,20,tr)
> 
> This didn't work - what does the @1000 indicate?


It shouldn't be there, If it's defined as 1000 in sip.conf make your
dial string

exten => 1000,1,Dial(SIP/1000,20,Ttr)

I don't know what the Tt does (lack of documentation) but adding an r
has asterisk generate the ringing (when dialing calls to outside
providers/cards) and m will insert music on hold in most cases.
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